FIG. 1 depicts a diagram of telecommunications system 100 in the prior art. Telecommunications system 100 comprises wireless telecommunications terminals 101 and 102; access points 103-1 through 103-M that serve basic service areas 110-1 through 110-M, respectively, wherein M is a positive integer; server 105; telecommunications terminal 106; and telecommunications network 120, interconnected as shown.
Telecommunications system 100 is capable of Session Initiation Protocol (or “SIP”) signaling. SIP is a standard protocol for initiating an interactive user session (i.e., a “call”) that involves multimedia elements such as voice, chat, video, and so forth.
Basic service area 110-m, for m=1 through M, is the service area in which shared access of other nodes in telecommunications system 100 is provided to telecommunications terminals such as terminals 101 and 102. As depicted in FIG. 1, basic service area 110-m is in an IEEE 802.11 wireless local area network. In area 110-m, the one or more wireless telecommunications terminals that make up the basic service set of area 110-m are able to access other nodes in system 100 via a shared-communications channel supported by access point 103-m. 
Telecommunications network 120 is a telecommunications network such as the Internet, the Public Switched Telephone Network (PSTN), and so forth. Network 120 comprises or is connected to one or more transmission-related nodes such as gateways, routers, or switches that are used to direct packets from one or more sources to the correct destinations of those packets.
The service provided by the path that links a first node with a second node is characterized by its “quality of service,” which, for the purposes of this specification, is defined as a function of the bandwidth, error rate, and latency from one node to another. For example, a shared-communications channel that links a wireless terminal such as terminal 101 with an access point such as access point 103-1 is subject to a quality-of-service level.
For the purposes of this specification, the “bandwidth” from one node to another is defined as an indication of the amount of information per unit time that can be transported from the first node to the second. Typically, bandwidth is measured in bits or bytes per second. For the purposes of this specification, the “error rate” from one node to another is defined as an indication of the amount of information that is corrupted as it travels from the first node to the second. Typically, error rate is measured in bit errors per number of bits transmitted or in packets lost per number of packets transmitted. For the purposes of this specification, the “latency” from one node to another is defined as an indication of how much time is required to transport information from one node to another. Typically, latency is measured in seconds.
Each of telecommunications terminals 101 and 102, as well as terminal 106, is a communications device such as a local area network telephone, a notebook computer, a personal digital assistant [PDA], a tablet PC, and so forth. Each telecommunications terminal has an associated contact identifier (e.g., telephone number, email address, Internet Protocol address, etc.) that uniquely identifies that terminal in the address space of telecommunications system 100. Terminals 101 and 102 communicate, through one or more access points 103-1 through 103-M, with other telecommunications terminals that have connectivity with network 120, such as terminal 106. For example, terminal 101 is presently associated with access point 103-1 as depicted in FIG. 1 and uses the corresponding shared-communications channel to communicate wirelessly with other devices. In order to communicate, a user at a first telecommunications terminal in system 100, such as terminal 101, places a “call” (e.g., voice call, email, text chat, video, etc.) to a user at a second terminal in system 100, such as terminal 106. Terminals 101 and 102 are also capable of communicating with each other with call control signaling being routed through one or more nodes connected to network 120, as described below.
Server 105 is a Session Initiation Protocol (SIP) proxy server that interacts with other nodes in setting up and managing calls; server 105's role in handling a call is represented by the prior-art message flow depicted in FIG. 2. In the message flow, the user of telecommunications terminal 101 is calling a user at telecommunications terminal 106. Terminal 101's user enters a command into terminal 101 to place a call (e.g., a voice telephone call, a video conference, a text-based instant message [IM], etc.) to the other user. Terminal 101 transmits message 201, a SIP INVITE message, to server 105. Message 201 comprises (i) a traffic stream description (e.g., a SIP Real-Time Protocol [RTP] payload type, etc.) that specifies the nature of the call and (ii) the user identifier of the person being called.
Server 105 determines which terminal is currently associated with the user identifier of the called user (i.e., the “destination terminal”), and transmits message 202 (e.g., a SIP OPTIONS message, etc.) to the destination terminal, in this case terminal 106, to determine the capabilities of that terminal (e.g., whether the destination terminal supports certain types of media, etc.).
Server 105 also transmits response message 203 (e.g., a SIP TRYING message, etc.) back to wireless terminal 101, indicating that the server is attempting to set up the call.
Destination terminal 106 transmits message 204 to server 105 (e.g., a SIP OK message, etc.), in response to message 202, and server 105 informs calling terminal 101 via message 205 that destination terminal 106 is able to participate in the call. Terminal 101 then acknowledges having received message 205 via messages 206 and 207, and terminals 101 and 106 start exchanging traffic packets in real-time protocol (RTP) stream 208.
Although the prior-art message flow in FIG. 2 depicts how a call is set up, what is missing from the message flow is how to determine whether to admit the call in the first place. Call admission control is necessary, considering that terminal 101 is also competing with other terminals in its basic service area (i.e., area 101-1) for the shared-communications channel provided by access point 103-1. In fact, each of access points 103-1 through 103-M has to be able to handle multiple traffic streams, each of which comprising a series of packets, that are transmitted to or from wireless terminals via the corresponding shared-communications channel. However, access point 103-m itself may or may not be capable of handling the requests from an associated wireless terminal to admit such a traffic stream. Even if access point 103-m provides call admission control, the added call admission functionality can add cost to each access point. Therefore, the need exists to provide the call admission control functionality that is necessary to handle multiple traffic streams on a shared-communications channel, while ensuring an acceptable quality-of-service level, without some of the disadvantages in the prior art.